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<% do_pagehead(); %>> <title><% nvram_get("router_name"); %> - SiPath Overview</title> <script type="text/javascript">document.title = "<% nvram_get("router_name"); %>" + sipath.titl; </script> </head> <body class="gui"> <div id="wrapper"> <div id="content"> <div id="header"> <div id="logo"><h1><% show_control(); %></h1></div> <% do_menu("Sipath.asp","Sipath.asp"); %> </div> <div id="main"> <div id="contents"> <h2>SIPath</h2> <p>Welcome to your <a href="http://sipath.sourceforge.net" target="new">SIP at Home</a> web interface for the <a href="http://sip-router.org/ser/" target="new">SIP Express Router</a> running on <a href="http://www.dd-wrt.com">DD-WRT Embedded Linux</a>.<br /> <b>Short SIP at Home HowTo:</b><br /> Standard configuration (/etc/ser/ser.cfg) will cause the router to function as an Outbound Proxy and local Registrar Server.<br /> To enable the first function simply configure the routers IP address as <i>Outbound Proxy</i> in your SIP phones configuration.<br /> Every registration your phone sends to your VoIP Provider will then pass through the SIP Express Router.<br /> At the same time such a registration is forwarded to your Provider the router will remember you as being registered and will route calls to registered phones locally, so no internet traffic is being produced. A successful local registration can be checked on the Phonebook page in the table <i>Contacts</i>. All listed <i>aor</i>s (Address Of Record) may be called locally.<br /> All authentication passwords of your SIP account are statelessly forwarded to the <i>SIP Server</i>.<br /> If a callees number cannot be resolved locally, the call signalling will be forwarded to the designated destination, being the <i>SIP Server</i> you set in the calling phones configuration. This may imply that your VoIP Service Provider configured in your SIP phones configuration as <i>SIP Server</i> (or similar), may be connected and an external call may be established.<br /> <br /> If you cannot place or receive calls from or to your phones over the Internet or if your phones ring but you cannot talk to someone outside your network as soon as you raise the receiver, this may indicate that your firewall settings prevent the establishment of a RTP stream. In such a case, please refer to your firewalls configuration howto or manual.<br /> <br /> <b>Examples:</b><br /> <u>John</u> has configured this routers IP as the Outbound Proxy in his SIP phone and did a reset. His username is <u>123456</u>, which is a registered account at an Internet VoIP Provider.<br /> <u>Peter</u> also changed the setting of the Outbound Proxy of his SIP phone and restarted it. He also has a registered user account which is <u>234567</u> at the same provider as John.<br /> <u>Chasey</u> has a user account at the same provider as John and Peters, yet she isn't connected to John and Peters SIP at Home Router. Chasey's username at the boys provider is <u>345678</u>.<br /> <u>Marilyn</u>, living on the other side of town, only has a standard landline phone with the number <u>456789</u>.<br /> <br /> 1. Scenario - Peter wants to call John:<br /> So Peter dials 123456 with his SIP phone. The phone at first connects the SIP at Home Router. The router checks if 123456 is locally registered, which is the case. Peters phones call-invitation is being routed locally to Johns phone which thereby may start to ring. A RTP media stream is established between the two phones directly.<br /> <br /> 2. Scenario - John wants to call Chasey:<br /> John therefore dials 345678. The phone uses the SIP at Home Router as the next signalling instance (Outbound Proxy) but 345678 is locally unknown and therefore the call is being forwarded to the <i>SIP Server</i> John has set in his phones configuration. This <i>SIP Server</i> checks the number which in this case matches with the user account of Chasey. The call invitation is then forwarded to Chaseys SIP phone. A RTP media stream is established between John's and Chasey's phone, either with the SIP Provider proxying the media or not, depending on the Providers settings.<br /> <br /> 3. Scenario - Peter wants to call Marilyn:<br /> So he dials 00 as the prefix for landline calls and then 456789. The phone uses the SIP at Home Router as Outbound Proxy but 00456789 is locally unknown so the call is forwarded to the <i>SIP Server</i> Peter set in his phones settings. This SIP Server recognises from the prefix that Peter wants to place a call into the PSTN and directs the call to its gateway which connects to Marilyns phone. A RTP media stream is established between Peters phone, the Providers PSTN-Gateway over the PSTN to Marilyns phone.<br /> <br /> Please note that this software is under development and comes with no warranty. </p><br /> </div> </div> <div id="helpContainer"> <div id="help"> <div><h2><% tran("share.help"); %></h2></div> <br /> <!--<a href="javascript:openHelpWindow<% nvram_selmatch("dist_type","micro","Ext"); %>('HSipath.asp')"><% tran("share.more"); %></a>--> </div> </div> <div id="floatKiller"></div> <% do_statusinfo(); %> </div> </div> </body></html>⌨️ 快捷键说明
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